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/*
* libjingle
* Copyright 2004--2011, Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_SESSION_PHONE_WEBRTCVOICEENGINE_H_
#define TALK_SESSION_PHONE_WEBRTCVOICEENGINE_H_
#include <map>
#include <set>
#include <string>
#include <vector>
#include "talk/base/buffer.h"
#include "talk/base/byteorder.h"
#include "talk/base/logging.h"
#include "talk/base/scoped_ptr.h"
#include "talk/base/stream.h"
#include "talk/session/phone/channel.h"
#include "talk/session/phone/rtputils.h"
#include "talk/session/phone/webrtccommon.h"
#include "talk/session/phone/webrtcvoe.h"
#ifdef WEBRTC_RELATIVE_PATH
#include "voice_engine/main/interface/voe_base.h"
#else
#include "third_party/webrtc/files/include/voe_base.h"
#endif // WEBRTC_RELATIVE_PATH
namespace cricket {
// WebRtcSoundclipStream is an adapter object that allows a memory stream to be
// passed into WebRtc, and support looping.
class WebRtcSoundclipStream : public webrtc::InStream {
public:
WebRtcSoundclipStream(const char* buf, size_t len)
: mem_(buf, len), loop_(true) {
}
void set_loop(bool loop) { loop_ = loop; }
virtual int Read(void* buf, int len);
virtual int Rewind();
private:
talk_base::MemoryStream mem_;
bool loop_;
};
// WebRtcMonitorStream is used to monitor a stream coming from WebRtc.
// For now we just dump the data.
class WebRtcMonitorStream : public webrtc::OutStream {
virtual bool Write(const void *buf, int len) {
return true;
}
};
class AudioDeviceModule;
class VoETraceWrapper;
class VoEWrapper;
class VoiceProcessor;
class WebRtcSoundclipMedia;
class WebRtcVoiceMediaChannel;
// WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
// It uses the WebRtc VoiceEngine library for audio handling.
class WebRtcVoiceEngine
: public webrtc::VoiceEngineObserver,
public webrtc::TraceCallback,
public webrtc::VoEMediaProcess {
public:
WebRtcVoiceEngine();
// Dependency injection for testing.
WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
VoEWrapper* voe_wrapper_sc,
VoETraceWrapper* tracing);
~WebRtcVoiceEngine();
bool Init();
void Terminate();
int GetCapabilities();
VoiceMediaChannel* CreateChannel();
SoundclipMedia* CreateSoundclip();
bool SetOptions(int options);
bool SetDevices(const Device* in_device, const Device* out_device);
bool GetOutputVolume(int* level);
bool SetOutputVolume(int level);
int GetInputLevel();
bool SetLocalMonitor(bool enable);
const std::vector<AudioCodec>& codecs();
bool FindCodec(const AudioCodec& codec);
bool FindWebRtcCodec(const AudioCodec& codec, webrtc::CodecInst* gcodec);
void SetLogging(int min_sev, const char* filter);
bool RegisterProcessor(uint32 ssrc,
VoiceProcessor* voice_processor,
MediaProcessorDirection direction);
bool UnregisterProcessor(uint32 ssrc,
VoiceProcessor* voice_processor,
MediaProcessorDirection direction);
// Method from webrtc::VoEMediaProcess
virtual void Process(const int channel,
const webrtc::ProcessingTypes type,
WebRtc_Word16 audio10ms[],
const int length,
const int sampling_freq,
const bool is_stereo);
// For tracking WebRtc channels. Needed because we have to pause them
// all when switching devices.
// May only be called by WebRtcVoiceMediaChannel.
void RegisterChannel(WebRtcVoiceMediaChannel *channel);
void UnregisterChannel(WebRtcVoiceMediaChannel *channel);
// May only be called by WebRtcSoundclipMedia.
void RegisterSoundclip(WebRtcSoundclipMedia *channel);
void UnregisterSoundclip(WebRtcSoundclipMedia *channel);
// Called by WebRtcVoiceMediaChannel to set a gain offset from
// the default AGC target level.
bool AdjustAgcLevel(int delta);
// Called by WebRtcVoiceMediaChannel to configure echo cancellation
// and noise suppression modes.
bool SetConferenceMode(bool enable);
VoEWrapper* voe() { return voe_wrapper_.get(); }
VoEWrapper* voe_sc() { return voe_wrapper_sc_.get(); }
int GetLastEngineError();
// Set the external ADMs. This can only be called before Init.
bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm,
webrtc::AudioDeviceModule* adm_sc);
// Check whether the supplied trace should be ignored.
bool ShouldIgnoreTrace(const std::string& trace);
private:
typedef std::vector<WebRtcSoundclipMedia *> SoundclipList;
typedef std::vector<WebRtcVoiceMediaChannel *> ChannelList;
struct CodecPref {
const char* name;
int clockrate;
int payload_type;
};
void Construct();
void ConstructCodecs();
bool InitInternal();
void ApplyLogging(const std::string& log_filter);
virtual void Print(const webrtc::TraceLevel level,
const char* trace_string, const int length);
virtual void CallbackOnError(const int channel, const int errCode);
// Given the device type, name, and id, find device id. Return true and
// set the output parameter rtc_id if successful.
bool FindWebRtcAudioDeviceId(
bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
bool FindChannelAndSsrc(int channel_num,
WebRtcVoiceMediaChannel** channel,
uint32* ssrc) const;
bool FindChannelNumFromSsrc(uint32 ssrc,
MediaProcessorDirection direction,
int* channel_num);
bool ChangeLocalMonitor(bool enable);
bool PauseLocalMonitor();
bool ResumeLocalMonitor();
// When a voice processor registers with the engine, it is connected
// to either the Rx or Tx signals, based on the direction parameter.
// SignalXXMediaFrame will be invoked for every audio packet.
sigslot::signal2<uint32, AudioFrame*> SignalRxMediaFrame;
sigslot::signal2<uint32, AudioFrame*> SignalTxMediaFrame;
static const int kDefaultLogSeverity = talk_base::LS_WARNING;
static const CodecPref kCodecPrefs[];
// The primary instance of WebRtc VoiceEngine.
talk_base::scoped_ptr<VoEWrapper> voe_wrapper_;
// A secondary instance, for playing out soundclips (on the 'ring' device).
talk_base::scoped_ptr<VoEWrapper> voe_wrapper_sc_;
talk_base::scoped_ptr<VoETraceWrapper> tracing_;
// The external audio device manager
webrtc::AudioDeviceModule* adm_;
webrtc::AudioDeviceModule* adm_sc_;
int log_level_;
std::string log_filter_;
bool is_dumping_aec_;
std::vector<AudioCodec> codecs_;
bool desired_local_monitor_enable_;
talk_base::scoped_ptr<WebRtcMonitorStream> monitor_;
SoundclipList soundclips_;
ChannelList channels_;
// channels_ can be read from WebRtc callback thread. We need a lock on that
// callback as well as the RegisterChannel/UnregisterChannel.
talk_base::CriticalSection channels_cs_;
webrtc::AgcConfig default_agc_config_;
bool initialized_;
talk_base::CriticalSection signal_media_critical_;
};
// WebRtcMediaChannel is a class that implements the common WebRtc channel
// functionality.
template <class T, class E>
class WebRtcMediaChannel : public T, public webrtc::Transport {
public:
WebRtcMediaChannel(E *engine, int channel)
: engine_(engine), voe_channel_(channel), sequence_number_(-1) {}
E *engine() { return engine_; }
int voe_channel() const { return voe_channel_; }
bool valid() const { return voe_channel_ != -1; }
protected:
// implements Transport interface
virtual int SendPacket(int channel, const void *data, int len) {
if (!T::network_interface_) {
return -1;
}
// We need to store the sequence number to be able to pick up
// the same sequence when the device is restarted.
// TODO: Remove when WebRtc has fixed the problem.
int seq_num;
if (!GetRtpSeqNum(data, len, &seq_num)) {
return -1;
}
if (sequence_number() == -1) {
LOG(INFO) << "WebRtcVoiceMediaChannel sends first packet seqnum="
<< seq_num;
}
sequence_number_ = seq_num;
talk_base::Buffer packet(data, len, kMaxRtpPacketLen);
return T::network_interface_->SendPacket(&packet) ? len : -1;
}
virtual int SendRTCPPacket(int channel, const void *data, int len) {
if (!T::network_interface_) {
return -1;
}
talk_base::Buffer packet(data, len, kMaxRtpPacketLen);
return T::network_interface_->SendRtcp(&packet) ? len : -1;
}
int sequence_number() const {
return sequence_number_;
}
private:
E *engine_;
int voe_channel_;
int sequence_number_;
};
// WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
// WebRtc Voice Engine.
class WebRtcVoiceMediaChannel
: public WebRtcMediaChannel<VoiceMediaChannel, WebRtcVoiceEngine> {
public:
explicit WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine);
virtual ~WebRtcVoiceMediaChannel();
virtual bool SetOptions(int options);
virtual bool SetRecvCodecs(const std::vector<AudioCodec> &codecs);
virtual bool SetSendCodecs(const std::vector<AudioCodec> &codecs);
virtual bool SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions);
virtual bool SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions);
virtual bool SetPlayout(bool playout);
bool PausePlayout();
bool ResumePlayout();
virtual bool SetSend(SendFlags send);
bool PauseSend();
bool ResumeSend();
virtual bool AddStream(uint32 ssrc);
virtual bool RemoveStream(uint32 ssrc);
virtual bool GetActiveStreams(AudioInfo::StreamList* actives);
virtual int GetOutputLevel();
virtual bool SetOutputScaling(uint32 ssrc, double left, double right);
virtual bool GetOutputScaling(uint32 ssrc, double* left, double* right);
virtual bool SetRingbackTone(const char *buf, int len);
virtual bool PlayRingbackTone(uint32 ssrc, bool play, bool loop);
virtual bool PressDTMF(int event, bool playout);
virtual void OnPacketReceived(talk_base::Buffer* packet);
virtual void OnRtcpReceived(talk_base::Buffer* packet);
virtual void SetSendSsrc(uint32 id);
virtual bool SetRtcpCName(const std::string& cname);
virtual bool Mute(bool mute);
virtual bool SetSendBandwidth(bool autobw, int bps) { return false; }
virtual bool GetStats(VoiceMediaInfo* info);
// Gets last reported error from WebRtc voice engine. This should be only
// called in response a failure.
virtual void GetLastMediaError(uint32* ssrc,
VoiceMediaChannel::Error* error);
bool FindSsrc(int channel_num, uint32* ssrc);
void OnError(uint32 ssrc, int error);
bool sending() const { return send_ != SEND_NOTHING; }
int GetChannelNum(uint32 ssrc);
protected:
int GetLastEngineError() { return engine()->GetLastEngineError(); }
int GetOutputLevel(int channel);
bool GetRedSendCodec(const AudioCodec& red_codec,
const std::vector<AudioCodec>& all_codecs,
webrtc::CodecInst* send_codec);
bool EnableRtcp(int channel);
bool ResetRecvCodecs(int channel);
bool SetPlayout(int channel, bool playout);
static uint32 ParseSsrc(const void* data, size_t len, bool rtcp);
static Error WebRtcErrorToChannelError(int err_code);
private:
// A -10dB gain adjustment is actually +10 in
// AgcConfig.targetLeveldBOv
static const int kMinus10DbAdjustment = 10;
bool ChangePlayout(bool playout);
bool ChangeSend(SendFlags send);
typedef std::map<uint32, int> ChannelMap;
talk_base::scoped_ptr<WebRtcSoundclipStream> ringback_tone_;
std::set<int> ringback_channels_; // channels playing ringback
talk_base::scoped_ptr<webrtc::CodecInst> send_codec_;
int channel_options_;
bool agc_adjusted_;
bool dtmf_allowed_;
bool desired_playout_;
bool playout_;
SendFlags desired_send_;
SendFlags send_;
ChannelMap mux_channels_; // for multiple sources
// mux_channels_ can be read from WebRtc callback thread. Accesses off the
// WebRtc thread must be synchronized with edits on the worker thread. Reads
// on the worker thread are ok.
mutable talk_base::CriticalSection mux_channels_cs_;
};
}
#endif // TALK_SESSION_PHONE_WEBRTCVOICEENGINE_H_